Update audio code to use size_t more correctly, webrtc/common_audio/ portion.

webrtc/common_audio/include/audio_util.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 51 matching lines @@
 void S16ToFloat(const int16_t* src, size_t size, float* dest);
 void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
 void FloatToFloatS16(const float* src, size_t size, float* dest);
 void FloatS16ToFloat(const float* src, size_t size, float* dest);
 
 // Deinterleave audio from |interleaved| to the channel buffers pointed to
 // by |deinterleaved|. There must be sufficient space allocated in the
 // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
 // per buffer).
 template <typename T>
-void Deinterleave(const T* interleaved, int samples_per_channel,
+void Deinterleave(const T* interleaved, size_t samples_per_channel,
                   int num_channels, T* const* deinterleaved) {
   for (int i = 0; i < num_channels; ++i) {
     T* channel = deinterleaved[i];
     int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel; ++j) {
+    for (size_t j = 0; j < samples_per_channel; ++j) {
       channel[j] = interleaved[interleaved_idx];
       interleaved_idx += num_channels;
     }
   }
 }
 
 // Interleave audio from the channel buffers pointed to by |deinterleaved| to
 // |interleaved|. There must be sufficient space allocated in |interleaved|
 // (|samples_per_channel| * |num_channels|).
 template <typename T>
-void Interleave(const T* const* deinterleaved, int samples_per_channel,
+void Interleave(const T* const* deinterleaved, size_t samples_per_channel,
                 int num_channels, T* interleaved) {
   for (int i = 0; i < num_channels; ++i) {
     const T* channel = deinterleaved[i];
     int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel; ++j) {
+    for (size_t j = 0; j < samples_per_channel; ++j) {
       interleaved[interleaved_idx] = channel[j];
       interleaved_idx += num_channels;
     }
   }
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_